aboutsummaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
Diffstat
-rw-r--r--channels/chan_sip.c53
-rw-r--r--configs/sip.conf.sample26
-rw-r--r--include/asterisk/rtp.h15
-rw-r--r--main/rtp.c52
4 files changed, 117 insertions, 29 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6b3be234c..80b009157 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -948,8 +948,6 @@ static struct sip_pvt {
time_t lastrtprx; /*!< Last RTP received */
time_t lastrtptx; /*!< Last RTP sent */
int rtptimeout; /*!< RTP timeout time */
- int rtpholdtimeout; /*!< RTP timeout when on hold */
- int rtpkeepalive; /*!< Send RTP packets for keepalive */
struct sockaddr_in recv; /*!< Received as */
struct in_addr ourip; /*!< Our IP */
struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
@@ -2593,17 +2591,21 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
if (dialog->rtp) {
ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
+ ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive);
+ /* Set Frame packetization */
+ ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
+ dialog->autoframing = peer->autoframing;
}
if (dialog->vrtp) {
ast_rtp_setdtmf(dialog->vrtp, 0);
ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
+ ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
+ ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
}
- /* Set Frame packetization */
- if (dialog->rtp) {
- ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
- dialog->autoframing = peer->autoframing;
- }
ast_string_field_set(dialog, peername, peer->username);
ast_string_field_set(dialog, authname, peer->username);
ast_string_field_set(dialog, username, peer->username);
@@ -2642,8 +2644,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
dialog->noncodeccapability &= ~AST_RTP_DTMF;
ast_string_field_set(dialog, context, peer->context);
dialog->rtptimeout = peer->rtptimeout;
- dialog->rtpholdtimeout = peer->rtpholdtimeout;
- dialog->rtpkeepalive = peer->rtpkeepalive;
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
dialog->maxcallbitrate = peer->maxcallbitrate;
@@ -4171,16 +4171,19 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_settos(p->rtp, global_tos_audio);
+ ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
+ ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive);
if (p->vrtp) {
ast_rtp_settos(p->vrtp, global_tos_video);
ast_rtp_setdtmf(p->vrtp, 0);
ast_rtp_setdtmfcompensate(p->vrtp, 0);
+ ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
+ ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
+ ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
}
if (p->udptl)
ast_udptl_settos(p->udptl, global_tos_audio);
- p->rtptimeout = global_rtptimeout;
- p->rtpholdtimeout = global_rtpholdtimeout;
- p->rtpkeepalive = global_rtpkeepalive;
p->maxcallbitrate = default_maxcallbitrate;
}
@@ -10184,6 +10187,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
ast_cli(fd, " T1 minimum: %d\n", global_t1min);
ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No");
ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No");
+ ast_cli(fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
ast_cli(fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
@@ -11598,6 +11602,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (bridgepvt->udptl) {
if (p->t38.state == T38_PEER_REINVITE) {
sip_handle_t38_reinvite(bridgepeer, p, 0);
+ ast_rtp_set_rtptimers_onhold(p->rtp);
+ if (p->vrtp)
+ ast_rtp_set_rtptimers_onhold(p->vrtp); /* Turn off RTP timers while we send fax */
} else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
@@ -14714,23 +14721,23 @@ restartsearch:
(sip->owner->_state == AST_STATE_UP) &&
!sip->redirip.sin_addr.s_addr) {
if (sip->lastrtptx &&
- sip->rtpkeepalive &&
- (t > sip->lastrtptx + sip->rtpkeepalive)) {
+ ast_rtp_get_rtpkeepalive(sip->rtp) &&
+ (t > sip->lastrtptx + ast_rtp_get_rtpkeepalive(sip->rtp))) {
/* Need to send an empty RTP packet */
sip->lastrtptx = time(NULL);
ast_rtp_sendcng(sip->rtp, 0);
}
if (sip->lastrtprx &&
- (sip->rtptimeout || sip->rtpholdtimeout) &&
- (t > sip->lastrtprx + sip->rtptimeout)) {
+ (ast_rtp_get_rtptimeout(sip->rtp) || ast_rtp_get_rtpholdtimeout(sip->rtp)) &&
+ (t > sip->lastrtprx + ast_rtp_get_rtptimeout(sip->rtp))) {
/* Might be a timeout now -- see if we're on hold */
struct sockaddr_in sin;
ast_rtp_get_peer(sip->rtp, &sin);
if (sin.sin_addr.s_addr ||
- (sip->rtpholdtimeout &&
- (t > sip->lastrtprx + sip->rtpholdtimeout))) {
+ (ast_rtp_get_rtpholdtimeout(sip->rtp) &&
+ (t > sip->lastrtprx + ast_rtp_get_rtpholdtimeout(sip->rtp)))) {
/* Needs a hangup */
- if (sip->rtptimeout) {
+ if (ast_rtp_get_rtptimeout(sip->rtp)) {
while (sip->owner && ast_channel_trylock(sip->owner)) {
ast_mutex_unlock(&sip->lock);
usleep(1);
@@ -14751,8 +14758,12 @@ restartsearch:
has already been requested and we don't want to
repeatedly request hangups
*/
- sip->rtptimeout = 0;
- sip->rtpholdtimeout = 0;
+ ast_rtp_set_rtptimeout(sip->rtp, 0);
+ ast_rtp_set_rtpholdtimeout(sip->rtp, 0);
+ if (sip->vrtp) {
+ ast_rtp_set_rtptimeout(sip->vrtp, 0);
+ ast_rtp_set_rtpholdtimeout(sip->vrtp, 0);
+ }
}
}
}
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 37e3ebba3..33959eff4 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -91,12 +91,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
-;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
- ; when we're not on hold. This is to be able to hangup
- ; a call in the case of a phone disappearing from the net,
- ; like a powerloss or grandma tripping over a cable.
-;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
- ; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
@@ -152,6 +146,21 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
;regcontext=sipregistrations
;
+;--------------------------- RTP timers ----------------------------------------------------
+; These timers are currently used for both audio and video streams. The RTP timeouts
+; are only applied to the audio channel.
+; The settings are settable in the global section as well as per device
+;
+;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're not on hold. This is to be able to hangup
+ ; a call in the case of a phone disappearing from the net,
+ ; like a powerloss or grandma tripping over a cable.
+;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
+ ; on the audio channel
+ ; when we're on hold (must be > rtptimeout)
+;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
+ ; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
@@ -196,8 +205,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
-; both parties have T38 support enabled in their Asterisk configuration (either general or
-; peer/user/friend sections)
+; both parties have T38 support enabled in their Asterisk configuration
+; This has to be enabled in the general section for all devices to work. You can then
+; disable it on a per device basis.
;
; t38pt_udptl = yes ; Default false
;
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index d7738b345..839648878 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -219,6 +219,21 @@ struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
int ast_rtp_codec_getformat(int pt);
+/*! \brief Set rtp timeout */
+void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
+/*! \brief Set rtp hold timeout */
+void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
+/*! \brief set RTP keepalive interval */
+void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
+/*! \brief Get RTP keepalive interval */
+int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
+/*! \brief Get rtp hold timeout */
+int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
+/*! \brief Get rtp timeout */
+int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
+/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
+void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
diff --git a/main/rtp.c b/main/rtp.c
index 53ef9f1c0..8761f5d5b 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -128,6 +128,11 @@ struct ast_rtp {
double rxtransit; /*!< Relative transit time for previous packet */
int lasttxformat;
int lastrxformat;
+
+ int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
+ int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
+ int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
+
/* DTMF Reception Variables */
char resp;
unsigned int lasteventendseqn;
@@ -521,6 +526,53 @@ unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
return interval;
}
+/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
+void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp)
+{
+ rtp->rtptimeout = (-1) * rtp->rtptimeout;
+ rtp->rtpholdtimeout = (-1) * rtp->rtpholdtimeout;
+}
+
+/*! \brief Set rtp timeout */
+void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout)
+{
+ rtp->rtptimeout = timeout;
+}
+
+/*! \brief Set rtp hold timeout */
+void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout)
+{
+ rtp->rtpholdtimeout = timeout;
+}
+
+/*! \brief set RTP keepalive interval */
+void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period)
+{
+ rtp->rtpkeepalive = period;
+}
+
+/*! \brief Get rtp timeout */
+int ast_rtp_get_rtptimeout(struct ast_rtp *rtp)
+{
+ if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
+ return 0;
+ return rtp->rtptimeout;
+}
+
+/*! \brief Get rtp hold timeout */
+int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp)
+{
+ if (rtp->rtptimeout < 0) /* We're not checking, but remembering the setting (during T.38 transmission) */
+ return 0;
+ return rtp->rtpholdtimeout;
+}
+
+/*! \brief Get RTP keepalive interval */
+int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp)
+{
+ return rtp->rtpkeepalive;
+}
+
void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
{
rtp->data = data;
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-01 08:55:18 +0000